The invention disclosed herein relates to voice-over-IP services and, more particularly, to improved methods for ensuring quality of service for voice communications using such services.
Packet telephony are services for providing voice traffic capabilities over a network, such as the Internet, a wide area network, an intranet, or a local area network, in which data is transmitted in packets. Voice-over-Internet Protocol (“VoIP”) services is a type of packet telephony in which the packets are carried by a network employing an IP routing protocol for routing traffic over the network. Current voice-over-IP services establish voice calls over data backbone networks by packetizing voice traffic into IP packets and transmitting them over the networks. Calls are first accepted based on a user profile and, once the destination is selected, all voice traffic is packetized in IP and transmitted.
In some VoIP systems, a gateway takes the voice communication from a traditional PBX or telephone switch, compresses it, packetizes it, converts it to IP format and routes it across the network to the destination. Another gateway at the destination receives the IP packet and reverses the operation to convert it back into the format needed for transmission to the receiving phone system. The gateways may be implemented as special hardware boxes or devices or as software modules running on network servers. The gateways work with gatekeeper devices or systems to identify the destination gateway and IP address of the identified recipient of the voice communication. Gatekeepers administer packet telephony call control and management, and administer call sessions to decide whether and how much bandwidth is dedicated to each requested communication.
For a voice call using VoIP to have acceptable quality, a number of Quality of Service (“QoS”) characteristics have to be met, such as minimum throughput, maximum delay, maximum delay variation, and packet loss. Service providers try to achieve these objectives by over-engineering and/or dedicating the IP network, which could be relatively expensive. On the other hand, mixing voice traffic with other traffic over the IP network, though cost effective, could significantly affect the quality of voice calls, especially during the busy hours.
There is therefore a need for a solution which guarantees the quality of a voice call on a VoIP network, once the call is launched.